I am running Asterisk 13.6.0 in an AWS instance, and I set it up with Twilio SIP trunk using pjsip_wizard.conf (nice feature!). I see that the calls actually reach the PBX, but for some reason, they are not caught by any of my extensions context.He..
When I do a SIP URI call from my softphone, the call is made directly to the destination host (p2p), bypassing the PBX. So I lose the possibility of recording, making statistics, etc …Is there a way to force URI calls through the PBX? I have fo..
I have a somewhat confusing use case.We use a mobile voip app and our users connect to our PBX via a public IP of our firewall which port forwards to asterisk (TLS and SRTP ports). Works fine. Sometimes however, our users are also connected to our ..
friends:I am facing cutoffs randomly when negotiating calls.The PBX dials the destination, the provider (softswitch) receives the request ** and sudenly the PBX hangs up the call* * while the provider is still dialing it, as a consequence the rem..
team! I’m planning to add fax functionality to my PBX. From research it seems that there is 2 options: spandsp and Digium. I lean towards Digium app, licensing is fine. However, they don’t have download for v13 Should I just download their vers..
all,Im new here and Im interested in building a small PBX with asterisk at home. I have one single PSTN line and ethernet cabling in place. I already have fairly decent PC that I can use (AMD FX 8350 16GB of RAM and RAID 10 SATA disks). I make and rece..